- 30 Nov, 2018 6 commits
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Matthew Waters authored
We already have our own versions so use them instead so that we don't have the potential for redefinition warnings between system GL headers and Qt's internal headers. Fixes gstreamer/gst-plugins-good#523
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- Consider GST_CLOCK_TIME_NONE as not to be used. - Complete "rtcp-feedback-retention-window" property getter/setter implementation.
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Closes #522
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- 29 Nov, 2018 3 commits
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Nicolas Dufresne authored
This is an extra internal recurisve lock use to avoid having to take both sink pad streams lock all the time. This patch renamed it INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream GST_PAD API.
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Nicolas Dufresne authored
This reverts "6f3734c3 rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock" and actually hold on the internal stream lock. This prevents in some needed case having a second streaming thread poping in and messing up event ordering.
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Nicolas Dufresne authored
This the first unit test of this element. It adds a test that verify that events are forwarded correctly.
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- 28 Nov, 2018 2 commits
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Pass riff codec-data as strf, not strd, which is where gst_riff_create_audio_caps() expects the WAVEFORMATEXTENSIBLE data. https://bugzilla.gnome.org/show_bug.cgi?id=757583 Fixes #234
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Jordan Petridіs authored
This is required before we enabled an indent test in the CI. gstreamer/gstreamer-project#33
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- 26 Nov, 2018 2 commits
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Thibault Saunier authored
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Thibault Saunier authored
Otherwise it might lead to deadlocks See https://gitlab.gnome.org/GNOME/pitivi/issues/2259 Closes #518
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- 24 Nov, 2018 2 commits
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While forwarding serialized event, we use gst_pad_forward() function. In the forward callback (GstPadForwardFunction) we always return TRUE. Returning true there will stop the dispatching procedure. As a side effect, only one events is receiving the events. This breaks when sending EOS from the applicaiton, it also breaks the latency tracer.
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Seungha Yang authored
Use build arguments consistent with core and -base. This can also remove noisy "C4819" warning of non-us locale MSVC.
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- 23 Nov, 2018 1 commit
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Xavier Claessens authored
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- 21 Nov, 2018 2 commits
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Nicolas Dufresne authored
The previous patch did not even compile on any possible platform or C standard. That commit also didn't have a proper commit message. Android ships Linux with a different signature for ioctl. They first released an ioctl with int as request type, and later "fixed" it by adding an override with unsign, which is still not matching Linux and BSD implementation which uses unsigned long int.
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Xavier Claessens authored
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- 19 Nov, 2018 2 commits
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Xavier Claessens authored
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PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it supports. We were previously exposing a maximum rate of INT_MAX, which is incorrect, but worked because nothing was really using a rate greater than 384000 kHz. While playing DSD data, we hit a case where there might be very high sample rates (>1MHz), and pulsesink fails during stream creation with such streams because it erroneously advertises that it supports such rates. Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in the caps string. Instead, we fix up the rate to what we actually support whenever we use our macro caps.
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- 15 Nov, 2018 1 commit
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This patch enables matroskademux to receive seeks before it reaches GST_MATROSKA_READ_STATE_DATA. Closes gstreamer/gst-plugins-good#514 This also enables receiving seeks in the element READY state. When such a seek is received, it is stored to be later handled when GST_MATROSKA_READ_STATE_DATA is reached.
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- 13 Nov, 2018 2 commits
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Reset RTPSession when rtpsession changes state from PAUSED to READY. Without this change, a stored last_rtptime in RTPSource could interfere with RTP timestamp generation in RTCP Sender Report. Fixes #510
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- 12 Nov, 2018 2 commits
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Mathieu Duponchelle authored
Fixes #516
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This commit adds a .gitlab-ci.yml file, which uses a feature to fetch the config from a centralized repository. The intent is to have all the gstreamer modules use the same configuration. The configuration is currently hosted at the gst-ci repository under the gitlab/ci_template.yml path. Part of gstreamer-project#29
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- 08 Nov, 2018 1 commit
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Nicolas Dufresne authored
If there was no interlace-mode field in the caps. Read back the value selected by the driver. This way, if the driver does not support progressive, then it will automatically negotiate the returned mode unless this mode is not supported by GStreamer. This method was already used for colorimetry. Just like colorimetry, the interlace mode is not longer probed by v4l2src dues to performance issues. Fixes #511
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- 07 Nov, 2018 3 commits
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Return the size / total duration as a ballpark estimate. gstreamer/gst-plugins-base#60
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Return the upstream size over the duration as a first estimate. gstreamer/gst-plugins-base#60
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Otherwise signal handlers from bindings will take ownership of them as they are still floating, and we won't own a reference inside rtpbin anymore. Fixes gstreamer/gst-plugins-good#515
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- 05 Nov, 2018 4 commits
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Decreasing timestamps break rtmpsink. With contributions from Olivier Crête. https://bugzilla.gnome.org/show_bug.cgi?id=796382
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Matthew Waters authored
Remove the git directory
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Haihao Xiang authored
This fixes gstreamer/gst-plugins-good#513
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- 01 Nov, 2018 1 commit
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If ctts (CompositionOffsetBox) has larger sample_offset (offset between PTS and DTS) than (2 * duration) of the stream, assume the ctts box to be corrupted and ignore the box. https://bugzilla.gnome.org/show_bug.cgi?id=797262
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- 28 Oct, 2018 6 commits
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Olivier Crête authored
This element doesn't support planar audio yet.
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Nirbheek Chauhan authored
Without these dependencies, the enumtype may not be generated when the test is built, which will cause a compile failure.
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Nirbheek Chauhan authored
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Olivier Crête authored
Make it possible to modify the SDES in a packet at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=763502
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